1. Field of the Invention
The present invention relates to electronic hearing aid devices for use by the hearing impaired and to methods for providing hearing compensation. More particularly, the present invention relates to such devices and methods utilizing both analog and digital signal processing techniques.
2. The Prior Art
One of the most common complaints made by hearing aid users is the inability to hear in the presence of noise. As a result, several researchers have opted for acoustic schemes which suppress noise to enhance the intelligibility of sound. Examples of this approach are found in U.S. Pat. No. 4,025,721 to Graupe, U.S. Pat. No. 4,405,831 to Michaelson, U.S. Pat. No. 4,185,168 to Graupe et al., U.S. Pat. No. 4,188,667 to Graupe et al., U.S. Pat. No. 4,025,721 to Graupe et al., U.S. Pat. No. 4,135,590 to Gaulder, and U.S. pat. No. 4,759,071 to Heide et al.
Other approaches have focussed upon feedback suppression and equalization (U.S. Pat. No. 4,602,337 to Cox, and U.S. Pat. No. 5,016,280 to Engebretson), dual microphone configurations (U.S. Pat. No. 4,622,440 to Slavin and U.S. Pat. No. 3,927,279 to Nakamura et al.), or upon coupling to the ear in unusual ways (e.g., RF links, electrical stimulation, etc.) to improve intelligibility. Examples of these approaches are found in U.S. Pat. No. 4,545,082 to Engebretson, U.S. Pat. No. 4,052,572 to Shafer, U.S. Pat. No. 4,852,177 to Ambrose, and U.S. Pat. No. 4,731,850 to Levitt.
Still other approaches have opted for digital programming control implementations which will accommodate a multitude of compression and filtering schemes. Examples of such approaches are found in U.S. Pat. No. 4,471,171 to Kopke et al. and U.S. Pat. No. 5,027,410 to Williamson. Some approaches, such as that disclosed in U.S. Pat. No. 5,083,312 to Newton, utilize hearing aid structures which allow flexibility by accepting control signals received remotely by the aid.
U.S. Pat. No. 4,187,413 to Moser discloses an approach for a digital hearing aid which uses an analog-to-digital converter, a digital-to-analog converter, and implements a fixed transfer function H(z). However, a review of neuro-psychological models in the literature and numerous measurements resulting in Steven's and Fechner's laws (see S. S. Stevens, Psychophysics, Wiley 1975; G. T. Fechner, Elemente der Psychophysik, Breitkopf u. Hartel, Leipzig, 1960) conclusively reveal that the response of the ear to input sound is nonlinear. Hence, no fixed transfer function H(z) exists which will fully compensate for hearing.
U.S. Pat. No. 4,425,481 to Mangold, et. al. discloses a programmable digital signal processing (DSP) device with features similar or identical to those commercially available, but with added digital control in the implementation of a three-band (lowpass, bandpass, and highpass) hearing aid. The outputs of the three frequency bands are each subjected to a digitally-controlled variable atttenuator, a limiter, and a final stage of digitally-controlled attenuation before being summed to provide an output. Control of attenuation is apparently accomplished by switching in response to different acoustic environments.
U.S. Pat. Nos. 4,366,349 and 4,419,544 to Adelman describe and trace the processing of the human auditory system, but do not reflect an understanding of the role of the outer hair cells within the ear as a muscle to amplify the incoming sound and provide increased basilar membrane displacement. These references assume that hearing deterioration makes it desirable to shift the frequencies and amplitude of the input stimulus, thereby transferring the location of the auditory response from a degraded portion of the ear to another area within the ear (on the basilar membrane) which has adequate response.
Mead C. Killion, The k-amp hearing aid: an attempt to present high fidelity for persons with impaired hearing, American Journal of Audiology, 2(2): pp. 52-74, July 1993, states that based upon the results of subjective listening tests for acoustic data processed with both linear gain and compression, either approach performs equally well. It is argued that the important factor in restoring hearing for individuals with losses is to provide the appropriate gain. Lacking a mathematically modeled analysis of that gain, several compression techniques have been proposed, e.g., U.S. Pat. No. 4,887,299 to Cummins; U.S. Pat. No. 3,920,931 to Yanick, Jr.; U.S. Pat. No. 4,118,604 to Yanick, Jr.; U.S. Pat. No. 4,052,571 to Gregory; U.S. Pat. No. 4,099,035 to Yanick, Jr. and U.S. Pat. No. 5,278,912 to Waldhauer. Some involve a linear fixed high gain at soft input sound levels and switch to a lower gain at moderate or loud sound levels. Others propose a linear gain at the soft sound intensities, a changing gain or compression at moderate intensities and a reduced, fixed linear gain at high or loud intensities. Still others propose table look-up systems with no details specified concerning formation of look-up tables, and others allow programmable gain without specification as to the operating parameters.
Switching between the gain mechanisms in each of these sound intensity regions has introduced significant distracting artifacts and distortion in the sound. Further, these gain-switched schemes have been applied typically in hearing aids to sound that is processed in two or three frequency bands, or in a single frequency band with pre-emphasis filtering.
Insight into the difficulty with prior art gain-switched schemes may be obtained by examining the human auditory system. For each frequency band where hearing has deviated from the normal threshold, a different sound compression is required to provide for normal hearing sensation to result. The application of gain schemes which attempt to combine more than a critical band (i.e., resolution band in hearing as defined in Jack Katz (Ed.) Handbook of Clinical Audiology, Williams & Wilkins, Baltimore, third ed., 1985) in frequency range cannot produce the appropriate hearing sensation in the listener. If, for example, it is desired to combine two frequency bands then some conditions must be met in order for the combination to correctly compensate for the hearing loss. These conditions for the frequency bands to be combined are that they have the same normal hearing threshold and dynamic range and require the same corrective hearing gain. In general, this does not occur even if a hearing loss is constant in amplitude across several critical bands of hearing. Failure to properly account for the adaptive full-range compression will result in degraded hearing or equivalently, loss of fidelity and intelligibility by the hearing impaired listener. Therefore, prior art which does not provide sufficient numbers of frequency bands to compensate for hearing losses will produce degraded hearing.
Several schemes have been proposed which use multiple bandpass filters followed by compression devices (see U.S. Pat. No. 4,396,806 to Anderson, U.S. Pat. No. 3,784,750 to Stearns et al., and U.S. Pat. No. 3,989,904 to Rohrer).
One example of prior art in U.S. Pat. No. 5,029,217 to Chabries focussed on an FFT frequency domain version of a human auditory model. The FFT implements an efficiently-calculated frequency domain filter which uses fixed filter bands in place of the critical band equivalents which naturally occur in the ear due to its unique geometry, thereby requiring that the frequency resolution of the FFT be equivalent to the smallest critical band to be compensated. The efficiency of the FFT is in large part negated by the fact that many additional filter bands are required in the FFT approach to cover the same frequency spectrum as a different implementation with critical bandwidth filters. This FFT implementation is complex and likely not suitable for low-power battery applications.
The prior-art FFT implementation introduces a block delay into the processing system inherent in the FFT itself. Blocks of samples are gathered for insertion into the FFT. This block delay introduces a time delay into the sound stream which is annoying and can induce stuttering when one tries to speak or can introduce a delay which sounds like an echo when low levels of compensation are required for the hearing impaired individual.
The prior art FFT implementation of a frequency-domain mapping between perceived sound and input sound levels for the normal and hearing impaired is undefined phenomenalogically. In other words, lacking a description of the perceived sound level versus input sound level for both the desired hearing response and the hearing impaired hearing response, these values were left to be measured.
For acoustic input levels below hearing (i.e. soft background sounds which are ever present), the FFT implementation described above provides excessive gain. This results in artifacts which add noise to the output signal. At hearing compensation levels greater than 60 dB, the processed background noise level can become comparable to the desired signal level in intensity thereby introducing distortion and reducing sound intelligibility.
As noted above, the hearing aid literature has proposed numerous solutions to the problem of hearing compensation for the hearing impaired. While the component parts that are required to assemble a high fidelity, full-range, adaptive compression system have been known since 1968, no one has to date proposed the application of the multiplicative AGC to the several bands of hearing to compensate for hearing losses. According to the present invention, this is precisely the operation required to provide near normal hearing perception to the hearing impaired.
As will be appreciated by those of ordinary skill in the art, there are three aspects to the realization of a high effectiveness aid for the hearing impaired. The first is the conversion of sound energy into electrical signals. The second is the processing of the electrical signals so as to compensate for the impairment of the particular individual. Finally, the processed electrical signals must be converted into sound energy in the ear canal.
Modern electret technology has allowed the construction of extremely small microphones with extremely high fidelity, thus providing a ready solution to the first aspect of the problem. The conversion of sound energy into electrical signals can be implemented with commercially available products. A unique solution to the problem of processing of the electrical signals to compensate for the impairment of the particular individual is set forth herein and in parent application Ser. No. 08/272,927 filed Jul. 8, 1994, now U.S. Pat. No. 5,500,902. The third aspect has, however, proved to be problematic, and is addressed by the present invention.
An in-the-ear hearing aid must operate on very low power and occupy only the space available in the ear canal. Because the hearing-impaired individual has lower sensitivity to sound energy than a normal individual, the hearing aid must deliver sound energy to the ear canal having an amplitude large enough to be heard and understood. The combination of these requirements dictates that the output transducer of the hearing aid must have high efficiency.
To meet this requirement transducer manufacturers such as Knowles have designed special iron-armature transducers that convert electrical energy into sound energy with high efficiency. To date this high efficiency has been achieved at the expense of extremely poor frequency response.
The frequency response of prior art transducers not only falls off well before the upper frequency limit of hearing, but also shows resonances starting at about 1 to 2 kHz, in a frequency range where they confound the information most useful in understanding human speech. These resonances are also primarily responsible for the feedback oscillation so commonly associated with hearing aids, and subject signals in the vicinity of the resonant frequencies to severe intermodulation distortion by mixing them with lower frequency signals. These resonances are a direct result of the mass of the iron armature, which is required to achieve good efficiency at low frequencies. In fact it is well known by those of ordinary skill in the art of transducer design that any transducer that is highly efficient at low frequencies will exhibit resonances in the mid-frequency range.
A counterpart to this problem occurs in high-fidelity loudspeaker design, and is solved in a universal manner by introducing two transducers, one that provides high efficiency transduction at low frequencies (a woofer), and one that provides high-quality transduction of the high frequencies (a tweeter). The audio signal is fed into a crossover network which directs the high frequency energy to the tweeter and the low frequency energy to the woofer. As will be appreciated by those of ordinary skill in the art, such a crossover network can be inserted either before or after power amplification.
In spite of its universal acceptance in high-fidelity audio systems, the two- speaker, crossover design has not found its way into commercial hearing aids.